linux-stable/sound/soc/soc-utils.c
Richard Fitzgerald 1ef34dd2b9
ASoC: soc-utils: Add helper to calculate BCLK from TDM info
Add a helper function snd_soc_tdm_params_to_bclk() to calculate
the bclk from params info and the tdm sots configuration.

When using a TDM frame of N slots of width W bits:

   bclk = sample_rate * N * W

As a convenience to simplify calling code, if the slot count or
slot width are 0 a value will be obtained from the params. This
allows calling code to use this one function to handle cases of
TDM where only one parameter is fixed, or I2S where the slot width
is fixed (for example to set a 32-bit slot for 24-bit samples).

Also as a convenience the slot count can optionally be rounded up
to a multiple. This is mainly useful for I2S systems, since I2S has
two phases of LRCLK the number of slots is always a multiple of 2.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220405135419.1230088-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2022-04-05 18:27:55 +01:00

271 lines
7.1 KiB
C

// SPDX-License-Identifier: GPL-2.0+
//
// soc-util.c -- ALSA SoC Audio Layer utility functions
//
// Copyright 2009 Wolfson Microelectronics PLC.
//
// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
// Liam Girdwood <lrg@slimlogic.co.uk>
#include <linux/platform_device.h>
#include <linux/export.h>
#include <linux/math.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots)
{
return sample_size * channels * tdm_slots;
}
EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size);
int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params)
{
int sample_size;
sample_size = snd_pcm_format_width(params_format(params));
if (sample_size < 0)
return sample_size;
return snd_soc_calc_frame_size(sample_size, params_channels(params),
1);
}
EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size);
int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots)
{
return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots);
}
EXPORT_SYMBOL_GPL(snd_soc_calc_bclk);
int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
{
int ret;
ret = snd_soc_params_to_frame_size(params);
if (ret > 0)
return ret * params_rate(params);
else
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);
/**
* snd_soc_tdm_params_to_bclk - calculate bclk from params and tdm slot info.
*
* Calculate the bclk from the params sample rate and the tdm slot count and
* tdm slot width. Either or both of tdm_width and tdm_slots can be 0.
*
* If tdm_width == 0 and tdm_slots > 0: the params_width will be used.
* If tdm_width > 0 and tdm_slots == 0: the params_channels will be used
* as the slot count.
* Both tdm_width and tdm_slots are 0: this is equivalent to calling
* snd_soc_params_to_bclk().
*
* If slot_multiple > 1 the slot count (or params_channels if tdm_slots == 0)
* will be rounded up to a multiple of this value. This is mainly useful for
* I2S mode, which has a left and right phase so the number of slots is always
* a multiple of 2.
*
* @params: Pointer to struct_pcm_hw_params.
* @tdm_width: Width in bits of the tdm slots.
* @tdm_slots: Number of tdm slots per frame.
* @slot_multiple: If >1 roundup slot count to a multiple of this value.
*
* Return: bclk frequency in Hz, else a negative error code if params format
* is invalid.
*/
int snd_soc_tdm_params_to_bclk(struct snd_pcm_hw_params *params,
int tdm_width, int tdm_slots, int slot_multiple)
{
if (!tdm_slots)
tdm_slots = params_channels(params);
if (slot_multiple > 1)
tdm_slots = roundup(tdm_slots, slot_multiple);
if (!tdm_width) {
tdm_width = snd_pcm_format_width(params_format(params));
if (tdm_width < 0)
return tdm_width;
}
return snd_soc_calc_bclk(params_rate(params), tdm_width, 1, tdm_slots);
}
EXPORT_SYMBOL_GPL(snd_soc_tdm_params_to_bclk);
static const struct snd_pcm_hardware dummy_dma_hardware = {
/* Random values to keep userspace happy when checking constraints */
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER,
.buffer_bytes_max = 128*1024,
.period_bytes_min = PAGE_SIZE,
.period_bytes_max = PAGE_SIZE*2,
.periods_min = 2,
.periods_max = 128,
};
static const struct snd_soc_component_driver dummy_platform;
static int dummy_dma_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
int i;
/*
* If there are other components associated with rtd, we shouldn't
* override their hwparams
*/
for_each_rtd_components(rtd, i, component) {
if (component->driver == &dummy_platform)
return 0;
}
/* BE's dont need dummy params */
if (!rtd->dai_link->no_pcm)
snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);
return 0;
}
static const struct snd_soc_component_driver dummy_platform = {
.open = dummy_dma_open,
};
static const struct snd_soc_component_driver dummy_codec = {
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
};
#define STUB_RATES SNDRV_PCM_RATE_8000_384000
#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
SNDRV_PCM_FMTBIT_U8 | \
SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_U16_LE | \
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S24_3LE | \
SNDRV_PCM_FMTBIT_U24_LE | \
SNDRV_PCM_FMTBIT_S32_LE | \
SNDRV_PCM_FMTBIT_U32_LE | \
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
/*
* Select these from Sound Card Manually
* SND_SOC_POSSIBLE_DAIFMT_CBP_CFP
* SND_SOC_POSSIBLE_DAIFMT_CBP_CFC
* SND_SOC_POSSIBLE_DAIFMT_CBC_CFP
* SND_SOC_POSSIBLE_DAIFMT_CBC_CFC
*/
static u64 dummy_dai_formats =
SND_SOC_POSSIBLE_DAIFMT_I2S |
SND_SOC_POSSIBLE_DAIFMT_RIGHT_J |
SND_SOC_POSSIBLE_DAIFMT_LEFT_J |
SND_SOC_POSSIBLE_DAIFMT_DSP_A |
SND_SOC_POSSIBLE_DAIFMT_DSP_B |
SND_SOC_POSSIBLE_DAIFMT_AC97 |
SND_SOC_POSSIBLE_DAIFMT_PDM |
SND_SOC_POSSIBLE_DAIFMT_GATED |
SND_SOC_POSSIBLE_DAIFMT_CONT |
SND_SOC_POSSIBLE_DAIFMT_NB_NF |
SND_SOC_POSSIBLE_DAIFMT_NB_IF |
SND_SOC_POSSIBLE_DAIFMT_IB_NF |
SND_SOC_POSSIBLE_DAIFMT_IB_IF;
static const struct snd_soc_dai_ops dummy_dai_ops = {
.auto_selectable_formats = &dummy_dai_formats,
.num_auto_selectable_formats = 1,
};
/*
* The dummy CODEC is only meant to be used in situations where there is no
* actual hardware.
*
* If there is actual hardware even if it does not have a control bus
* the hardware will still have constraints like supported samplerates, etc.
* which should be modelled. And the data flow graph also should be modelled
* using DAPM.
*/
static struct snd_soc_dai_driver dummy_dai = {
.name = "snd-soc-dummy-dai",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 384,
.rates = STUB_RATES,
.formats = STUB_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 384,
.rates = STUB_RATES,
.formats = STUB_FORMATS,
},
.ops = &dummy_dai_ops,
};
int snd_soc_dai_is_dummy(struct snd_soc_dai *dai)
{
if (dai->driver == &dummy_dai)
return 1;
return 0;
}
int snd_soc_component_is_dummy(struct snd_soc_component *component)
{
return ((component->driver == &dummy_platform) ||
(component->driver == &dummy_codec));
}
static int snd_soc_dummy_probe(struct platform_device *pdev)
{
int ret;
ret = devm_snd_soc_register_component(&pdev->dev,
&dummy_codec, &dummy_dai, 1);
if (ret < 0)
return ret;
ret = devm_snd_soc_register_component(&pdev->dev, &dummy_platform,
NULL, 0);
return ret;
}
static struct platform_driver soc_dummy_driver = {
.driver = {
.name = "snd-soc-dummy",
},
.probe = snd_soc_dummy_probe,
};
static struct platform_device *soc_dummy_dev;
int __init snd_soc_util_init(void)
{
int ret;
soc_dummy_dev =
platform_device_register_simple("snd-soc-dummy", -1, NULL, 0);
if (IS_ERR(soc_dummy_dev))
return PTR_ERR(soc_dummy_dev);
ret = platform_driver_register(&soc_dummy_driver);
if (ret != 0)
platform_device_unregister(soc_dummy_dev);
return ret;
}
void __exit snd_soc_util_exit(void)
{
platform_driver_unregister(&soc_dummy_driver);
platform_device_unregister(soc_dummy_dev);
}